Hola
Desde hace 2 semanas he tenido problemas con asterisk y telmex (ahora claro), primero los datos fueron borrados de la base de datos de telmex, luego de 2 semanas, se pudo recuperar la contraseña, ahora con los nuevos datos se conecta desde el softphone de telmex ( recibe y salen llamadas) pero desde asterisk No, aparece como registrado pero no salen llamadas, si entran luego de timbrar alrededor de 7 veces
Gracias
El log de los mensajes SIP con telmex intentando de llamar al 5818181 en Bogota.
La respuesta de Telmex es "Trying" (Intentando) pero como devuelve un ID de llamada, cuando cuelgo y que asterix quiere terminar la llamada, telmex responde que esa llamada no existe....
[2012-07-31 16:03:46] VERBOSE[8846] chan_sip.c: Reliably Transmitting (NAT) to 190.144.159.138:5060:
INVITE sip:5818181@190.144.159.138 SIP/2.0
Via: SIP/2.0/UDP <Mi-IP>:5060;branch=z9hG4bK364d7193;rport
Max-Forwards: 70
From: "18051917" <sip:00018051917@190.144.159.138>;tag=as42ae695f
To: <sip:5818181@190.144.159.138>
Contact: <sip:00018051917@<Mi-IP>:5060>
Call-ID: 189dad411e653cdd4597a7d224afe91e@<Mi-IP>:5060
CSeq: 102 INVITE
User-Agent: vincent
Date: Tue, 31 Jul 2012 21:03:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 180
v=0
o=root 528298341 528298341 IN IP4 <Mi-IP>
s=Asterisk PBX 1.8.7.1
c=IN IP4 <Mi-IP>
t=0 0
m=audio 11004 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
---
[2012-07-31 16:03:46] VERBOSE[8846] app_dial.c: -- Called SIP/Telmex(out)/5818181
[2012-07-31 16:03:46] VERBOSE[8847] app_mixmonitor.c: == Begin MixMonitor Recording SIP/109-0000003f
[2012-07-31 16:03:46] VERBOSE[7646] chan_sip.c:
<--- SIP read from UDP:190.144.159.138:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <Mi-IP>:5060;branch=z9hG4bK364d7193;rport
From: "18051917" <sip:00018051917@190.144.159.138>;tag=as42ae695f
To: <sip:5818181@190.144.159.138>
Call-ID: 189dad411e653cdd4597a7d224afe91e@<Mi-IP>:5060
CSeq: 102 INVITE
<------------->
[2012-07-31 16:03:46] VERBOSE[7646] chan_sip.c: --- (6 headers 0 lines) ---
[2012-07-31 16:04:40] NOTICE[7646] chan_sip.c: Peer '204' is now Reachable. (9ms / 2000ms)
[2012-07-31 16:04:44] VERBOSE[7646] chan_sip.c: -- Registered SIP '204' at 192.168.50.170:52139
[2012-07-31 16:04:58] VERBOSE[7625] chan_sip.c: == Extension Changed 109[ext-local] new state Idle for Notify User 102
[2012-07-31 16:04:58] VERBOSE[7625] chan_sip.c: == Extension Changed 109[ext-local] new state Idle for Notify User 132
[2012-07-31 16:04:58] VERBOSE[7625] chan_sip.c: == Extension Changed 109[ext-local] new state Idle for Notify User 126
[2012-07-31 16:04:58] VERBOSE[7625] chan_sip.c: == Extension Changed 109[ext-local] new state Idle for Notify User 130
[2012-07-31 16:04:58] VERBOSE[7625] chan_sip.c: == Extension Changed 109[ext-local] new state Idle for Notify User 121
[2012-07-31 16:04:58] VERBOSE[8846] chan_sip.c: Scheduling destruction of SIP dialog '189dad411e653cdd4597a7d224afe91e@<Mi-IP>:5060' in 32000 ms (Method: INVITE)
[2012-07-31 16:04:58] VERBOSE[8846] chan_sip.c: Reliably Transmitting (NAT) to 190.144.159.138:5060:
CANCEL sip:5818181@190.144.159.138 SIP/2.0
Via: SIP/2.0/UDP <Mi-IP>:5060;branch=z9hG4bK364d7193;rport
Max-Forwards: 70
From: "18051917" <sip:00018051917@190.144.159.138>;tag=as42ae695f
To: <sip:5818181@190.144.159.138>
Call-ID: 189dad411e653cdd4597a7d224afe91e@<Mi-IP>:5060
CSeq: 102 CANCEL
User-Agent: vincent
Content-Length: 0
---
[2012-07-31 16:04:58] VERBOSE[8846] chan_sip.c: Scheduling destruction of SIP dialog '189dad411e653cdd4597a7d224afe91e@<Mi-IP>:5060' in 32000 ms (Method: INVITE)
[2012-07-31 16:04:58] VERBOSE[8846] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/109-0000003f' in macro 'dialout-trunk'
[2012-07-31 16:04:58] VERBOSE[8846] pbx.c: == Spawn extension (from-internal, 95818181, 5) exited non-zero on 'SIP/109-0000003f'
[2012-07-31 16:04:58] VERBOSE[8846] pbx.c: -- Executing [h@from-internal:1] Hangup("SIP/109-0000003f", "") in new stack
[2012-07-31 16:04:58] VERBOSE[8846] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/109-0000003f'
[2012-07-31 16:04:58] VERBOSE[8847] app_mixmonitor.c: == End MixMonitor Recording SIP/109-0000003f
[2012-07-31 16:04:58] VERBOSE[7646] chan_sip.c:
<--- SIP read from UDP:190.144.159.138:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP <Mi-IP>:5060;branch=z9hG4bK364d7193;rport
From: "18051917" <sip:00018051917@190.144.159.138>;tag=as42ae695f
To: <sip:5818181@190.144.159.138>;tag=aprqngfrt-a65k3k10000c6
Call-ID: 189dad411e653cdd4597a7d224afe91e@<Mi-IP>:5060
CSeq: 102 CANCEL
<------------->
[2012-07-31 16:04:58] VERBOSE[7646] chan_sip.c: --- (6 headers 0 lines) ---
[2012-07-31 16:04:58] WARNING[7646] chan_sip.c: Remote host can't match request CANCEL to call '189dad411e653cdd4597a7d224afe91e@<Mi-IP>:5060'. Giving up.